--- BasiliskII/src/AmigaOS/audio_amiga.cpp 2000/04/10 18:52:33 1.7 +++ BasiliskII/src/AmigaOS/audio_amiga.cpp 2008/01/01 09:40:31 1.12 @@ -1,7 +1,7 @@ /* * audio_amiga.cpp - Audio support, AmigaOS implementation using AHI * - * Basilisk II (C) 1997-2000 Christian Bauer + * Basilisk II (C) 1997-2008 Christian Bauer * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -23,8 +23,11 @@ #include #include #include +#define __USE_SYSBASE #include #include +#include +#include #include "cpu_emulation.h" #include "main.h" @@ -36,14 +39,7 @@ #define DEBUG 0 #include "debug.h" - -// Supported sample rates, sizes and channels -int audio_num_sample_rates = 1; -uint32 audio_sample_rates[] = {22050 << 16}; -int audio_num_sample_sizes = 1; -uint16 audio_sample_sizes[] = {16}; -int audio_num_channel_counts = 1; -uint16 audio_channel_counts[] = {2}; +#define D1(x) ; // Global variables @@ -55,23 +51,40 @@ static int play_buf = 0; // Number static long sound_buffer_size; // Size of one audio buffer in bytes static int audio_block_fetched = 0; // Number of audio blocks fetched by interrupt routine +static bool main_mute = false; +static bool speaker_mute = false; +static ULONG supports_volume_changes = false; +static ULONG supports_stereo_panning = false; +static ULONG current_main_volume; +static ULONG current_speaker_volume; + // Prototypes static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/); +void audio_set_sample_rate_byval(uint32 value); +void audio_set_sample_size_byval(uint32 value); +void audio_set_channels_byval(uint32 value); /* * Initialization */ +// Set AudioStatus to reflect current audio stream format +static void set_audio_status_format(int sample_rate_index) +{ + AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; + AudioStatus.sample_size = audio_sample_sizes[0]; + AudioStatus.channels = audio_channel_counts[0]; +} + void AudioInit(void) { sample[0].ahisi_Address = sample[1].ahisi_Address = NULL; // Init audio status and feature flags - AudioStatus.sample_rate = audio_sample_rates[0]; - AudioStatus.sample_size = audio_sample_sizes[0]; - AudioStatus.channels = audio_channel_counts[0]; + audio_channel_counts.push_back(2); +// set_audio_status_format(); AudioStatus.mixer = 0; AudioStatus.num_sources = 0; audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut; @@ -106,6 +119,43 @@ void AudioInit(void) return; } + ULONG max_channels, sample_rate, frequencies, sample_rate_index; + + AHI_GetAudioAttrs(ahi_id, ahi_ctrl, + AHIDB_MaxChannels, (ULONG) &max_channels, + AHIDB_Frequencies, (ULONG) &frequencies, + TAG_END); + + D(bug("AudioInit: max_channels=%ld frequencies=%ld\n", max_channels, frequencies)); + + for (int n=0; n> 3) * AudioStatus.channels * audio_frames_per_block; @@ -119,12 +169,15 @@ void AudioInit(void) sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR); if (sample[0].ahisi_Address == NULL || sample[1].ahisi_Address == NULL) return; + AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl); AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl); // Set parameters play_buf = 0; - AHI_SetVol(0, 0x10000, 0x8000, ahi_ctrl, AHISF_IMM); + current_main_volume = current_speaker_volume = 0x10000; + AHI_SetVol(0, current_speaker_volume, 0x8000, ahi_ctrl, AHISF_IMM); + AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM); AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM); @@ -179,32 +232,66 @@ static __saveds __attribute__((regparm(3 play_buf ^= 1; // New buffer available? - if (audio_block_fetched) { + if (audio_block_fetched) + { audio_block_fetched--; - // Get size of audio data - uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo); - if (apple_stream_info) { - int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels; - D(bug("stream: work_size %d\n", work_size)); - if (work_size > sound_buffer_size) - work_size = sound_buffer_size; - - // Put data into AHI buffer (convert 8-bit data unsigned->signed) - if (AudioStatus.sample_size == 16) - Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size); - else { - uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)); - uint32 *q = (uint32 *)sample[play_buf].ahisi_Address; - int r = work_size >> 2; - while (r--) - *q++ = *p++ ^ 0x80808080; + if (main_mute || speaker_mute) + { + memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); + } + else + { + // Get size of audio data + uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo); + if (apple_stream_info) { + int32 sample_count = ReadMacInt32(apple_stream_info + scd_sampleCount); + + uint32 num_channels = ReadMacInt16(apple_stream_info + scd_numChannels); + uint32 sample_size = ReadMacInt16(apple_stream_info + scd_sampleSize); + uint32 sample_rate = ReadMacInt32(apple_stream_info + scd_sampleRate); + + D(bug("stream: sample_count=%ld num_channels=%ld sample_size=%ld sample_rate=%ld\n", sample_count, num_channels, sample_size, sample_rate >> 16)); + + // Yes, this can happen. + if(sample_count != 0) { + if(sample_rate != AudioStatus.sample_rate) { + audio_set_sample_rate_byval(sample_rate); + } + if(num_channels != AudioStatus.channels) { + audio_set_channels_byval(num_channels); + } + if(sample_size != AudioStatus.sample_size) { + audio_set_sample_size_byval(sample_size); + } + } + + if (sample_count < 0) + sample_count = 0; + + int work_size = sample_count * num_channels * (sample_size>>3); + D(bug("stream: work_size=%ld sound_buffer_size=%ld\n", work_size, sound_buffer_size)); + + if (work_size > sound_buffer_size) + work_size = sound_buffer_size; + + // Put data into AHI buffer (convert 8-bit data unsigned->signed) + if (AudioStatus.sample_size == 16) + Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size); + else { + uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)); + uint32 *q = (uint32 *)sample[play_buf].ahisi_Address; + int r = work_size >> 2; + while (r--) + *q++ = *p++ ^ 0x80808080; + } + if (work_size != sound_buffer_size) + memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size); } - if (work_size != sound_buffer_size) - memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size); } - } else + } + else memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); // Play next buffer @@ -212,7 +299,7 @@ static __saveds __attribute__((regparm(3 // Trigger audio interrupt to get new buffer if (AudioStatus.num_sources) { - D(bug("stream: triggering irq\n")); + D1(bug("stream: triggering irq\n")); SetInterruptFlag(INTFLAG_AUDIO); TriggerInterrupt(); } @@ -226,7 +313,7 @@ static __saveds __attribute__((regparm(3 void AudioInterrupt(void) { - D(bug("AudioInterrupt\n")); + D1(bug("AudioInterrupt\n")); // Get data from apple mixer if (AudioStatus.mixer) { @@ -234,13 +321,13 @@ void AudioInterrupt(void) r.a[0] = audio_data + adatStreamInfo; r.a[1] = AudioStatus.mixer; Execute68k(audio_data + adatGetSourceData, &r); - D(bug(" GetSourceData() returns %08lx\n", r.d[0])); + D1(bug(" GetSourceData() returns %08lx\n", r.d[0])); } else WriteMacInt32(audio_data + adatStreamInfo, 0); // Signal stream function audio_block_fetched++; - D(bug("AudioInterrupt done\n")); + D1(bug("AudioInterrupt done\n")); } @@ -250,16 +337,79 @@ void AudioInterrupt(void) * It is guaranteed that AudioStatus.num_sources == 0 */ -void audio_set_sample_rate(int index) +void audio_set_sample_rate_byval(uint32 value) { + bool changed = (AudioStatus.sample_rate != value); + if(changed) + { + ULONG sample_rate_index; + + // get index of sample rate closest to Hz + AHI_GetAudioAttrs(ahi_id, ahi_ctrl, + AHIDB_IndexArg, value >> 16, + AHIDB_Index, (ULONG) &sample_rate_index, + TAG_END); + + D(bug(" audio_set_sample_rate_byval requested rate=%ld Hz\n", value >> 16)); + + AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; + + AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, 0); + } + + D(bug(" audio_set_sample_rate_byval rate=%ld Hz\n", AudioStatus.sample_rate >> 16)); } -void audio_set_sample_size(int index) +void audio_set_sample_size_byval(uint32 value) { + bool changed = (AudioStatus.sample_size != value); + if(changed) { +// AudioStatus.sample_size = value; +// update_sound_parameters(); +// WritePrivateProfileInt( "Audio", "SampleSize", AudioStatus.sample_size, ini_file_name ); + } + D(bug(" audio_set_sample_size_byval %d\n", AudioStatus.sample_size)); } -void audio_set_channels(int index) +void audio_set_channels_byval(uint32 value) { + bool changed = (AudioStatus.channels != value); + if(changed) { +// AudioStatus.channels = value; +// update_sound_parameters(); +// WritePrivateProfileInt( "Audio", "Channels", AudioStatus.channels, ini_file_name ); + } + D(bug(" audio_set_channels_byval %d\n", AudioStatus.channels)); +} + +bool audio_set_sample_rate(int index) +{ + if(index >= 0 && index < audio_sample_rates.size() ) { + audio_set_sample_rate_byval( audio_sample_rates[index] ); + D(bug(" audio_set_sample_rate index=%ld rate=%ld\n", index, AudioStatus.sample_rate >> 16)); + } + + return true; +} + +bool audio_set_sample_size(int index) +{ + if(index >= 0 && index < audio_sample_sizes.size() ) { + audio_set_sample_size_byval( audio_sample_sizes[index] ); + D(bug(" audio_set_sample_size %d,%d\n", index,AudioStatus.sample_size)); + } + + return true; +} + +bool audio_set_channels(int index) +{ + if(index >= 0 && index < audio_channel_counts.size() ) { + audio_set_channels_byval( audio_channel_counts[index] ); + D(bug(" audio_set_channels %d,%d\n", index,AudioStatus.channels)); + } + + return true; } @@ -271,36 +421,95 @@ void audio_set_channels(int index) bool audio_get_main_mute(void) { - return false; + D(bug("audio_get_main_mute: mute=%ld\n", main_mute)); + + return main_mute; } uint32 audio_get_main_volume(void) { + D(bug("audio_get_main_volume\n")); + + ULONG volume = current_main_volume >> 8; // 0x10000 => 0x100 + + D(bug("audio_get_main_volume: volume=%08lx\n", volume)); + + return (volume << 16) + volume; + return 0x01000100; } bool audio_get_speaker_mute(void) { - return false; + D(bug("audio_get_speaker_mute: mute=%ld\n", speaker_mute)); + + return speaker_mute; } uint32 audio_get_speaker_volume(void) { + D(bug("audio_get_speaker_volume: \n")); + + if (audio_open) + { + ULONG volume = current_speaker_volume >> 8; // 0x10000 => 0x100 + + D(bug("audio_get_speaker_volume: volume=%08lx\n", volume)); + + return (volume << 16) + volume; + } + return 0x01000100; } void audio_set_main_mute(bool mute) { + D(bug("audio_set_main_mute: mute=%ld\n", mute)); + + if (mute != main_mute) + { + main_mute = mute; + } } void audio_set_main_volume(uint32 vol) { + D(bug("audio_set_main_volume: vol=%08lx\n", vol)); + + if (audio_open && supports_volume_changes) + { + ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); + + D(bug("audio_set_main_volume: volume=%08lx\n", volume)); + + current_main_volume = volume; + + AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); + } } void audio_set_speaker_mute(bool mute) { + D(bug("audio_set_speaker_mute: mute=%ld\n", mute)); + + if (mute != speaker_mute) + { + speaker_mute = mute; + } } void audio_set_speaker_volume(uint32 vol) { + D(bug("audio_set_speaker_volume: vol=%08lx\n", vol)); + + if (audio_open && supports_volume_changes) + { + ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); + + D(bug("audio_set_speaker_volume: volume=%08lx\n", volume)); + + current_speaker_volume = volume; + + AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); + } }